[cisco-voip] RTP only one way?

[Ryan Ratliff]

I don't know how this applies to SIP but if your AS has more than one  
IP address you should configure 'h323 bind voip srcaddr' under the  
appropriate interface.  Could very well be that the AS is telling the  
SIP device to send RTP to an IP address it can't route to.

HTH

-Ryan
On Apr 29, 2005, at 11:59 AM, Gerd Feiner wrote:

Hi Kevin,

no there aren't any routing issues.  The AS can ping the sipura and  
there are indeed RTP-packets from the AS to the sipura - about 1 every  
two seconds, while there are many many packtes from the sipura to the  
AS5350 ... when debugging SIP the AS5350 also tells about opening a  
recv-only audio-stream:

Apr 29 12:41:56 x.x.x.x 42940: Apr 29 10:48:30.444: sipSPIAddStream:  
AddStream in idle state to open a 'recvonly' media session

this is why I digged into and found that rtp send-receive command and  
its exactly what happens:  voip can speak and is heard on pstn, but not  
vice versa.

any ideas?

Brgds
Gerd

Am 29.04.2005 um 15:47 schrieb Kevin Thorngren:

> Hi Gerd,
>
> Typically one way voice issues are due to IP Routing issues.  Can the  
> 5300 ping the SIP Phone?
>
> This URL should help you diagnose the problem.
> http://www.cisco.com/en/US/tech/tk652/tk698/ 
> technologies_tech_note09186a008009484b.shtml
>
> Kevin
> On Apr 29, 2005, at 6:49 AM, Gerd Feiner wrote:
>
>> Hi there,
>>
>> we have an AS5350 and using as a SIP-Gateway to the PSTN.  Now, there  
>> is an intriguing issue:  SIP -> PSTN voice is audible, and there is  
>> an RTP stream from the SIP-device - SIPURA - to the mediagateway.   
>> But there is no stream in the SIP direction coming from the AS5350.   
>> I already found the
>>
>> voice rtp send-receive
>>
>> command - but it didn't do the trick. As of now, I wasn't able to  
>> ascertain the source of the problem.  It doesn't matter who is  
>> initiating the call, it's always the same effect.
>>
>> Don't know which part of our config you need, but here are a few:
>>
>> voice rtp send-recv
>> !
>> voice service pots
>>  fax protocol pass-through g711alaw
>> !
>> voice service voip
>>  signaling forward rawmsg
>>  fax protocol pass-through g711alaw
>>  sip
>>   rel1xx disable
>>   no call service stop
>> !
>> ...
>> !
>> interface Serial3/0:15
>>  no ip address
>>  isdn switch-type primary-net5
>>  isdn incoming-voice modem
>>  isdn sending-complete
>>  no cdp enable
>> !
>> voice-port 3/0:D
>>  bearer-cap Speech
>> !
>> dial-peer voice 1 pots
>>  tone ringback alert-no-PI
>>  application session
>>  incoming called-number 143677..
>>  destination-pattern .
>>  translate-outgoing calling 20
>>  translate-outgoing called 20
>>  supplementary-service pass-through
>>  no digit-strip
>>  direct-inward-dial
>>  port 3/0:D
>> !
>> dial-peer voice 2 voip
>>  tone ringback alert-no-PI
>>  application session
>>  incoming called-number .
>>  destination-pattern 143677..
>>  voice-class codec 10
>>  session protocol sipv2
>>  session target ipv4:x.x.x.x
>>  supplementary-service pass-through
>> !
>> !
>> dial-peer search type voice data
>> sip-ua
>>  nat symmetric check-media-src
>>  sip-server ipv4:x.x.x.x
>>
>> this isn't by far complete, but it seems to be the important part as  
>> I figured.  In addition, I don't really understand all of the  
>> commands set, most of it was from an example, part is from the ?-help  
>> system and another part is from cisco's voice config guide ...
>>
>> Glad if someone could help.
>>
>> Brgds,
>> Gerd Feiner
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
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